US 7584095 B2 Abstract An enhanced analysis-by-synthesis waveform interpolative speech coder able to operate at 2.8 kbps. Novel features include dual-predictive analysis-by-synthesis quantization of the slowly-evolving waveform, efficient parametrization of the rapidly-evolving waveform magnitude, and analysis-by-synthesis vector quantization of the rapidly evolving waveform parameter. Subjective quality tests indicate that it exceeds G.723.1 at 5.3 kbps, and of G.723.1 at 6.3 kbps.
Claims(18) 1. A computerized system for coding a digital input signal representative of an audio waveform, comprising:
a vector quantizer module comprising an input accepting a vector or rapidly evolving waveform spectra determined form the digital input signal;
a computer readable codebook comprising a plurality of arrays of quantized parametric vectors;
computer readable program instructions for searching said parameter codebook for one of said plurality of arrays of quantized parametric vectors which minimizes distortion; and
computer program instructions for outputting a computer readable indicator that points to the one of said plurality of arrays of quantized parameter vectors which minimizes distortion.
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10. A computer implemented method for coding a digital input signal representative of an audio waveform, comprising:
determining a vector of rapidly evolving waveform spectra from the digital input signal;
searching a computer readable codebook comprising a plurality of arrays of quantized parametric vector for one of said plurality of arrays of quantized parameter vectors which minimizes distortion; and
outputting a computer readable indicator that points to the one of said plurality of arrays of quantized parameter vectors which minimizes said distortion.
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Description This application claims the benefit of Provisional Patent Application No. 60/190,371 filed Mar. 17, 2000, which application is herein incorporated by reference. This application is a divisional of U.S. patent application Ser. No. 09/811,187, filed Mar. 16, 2001 now U.S. Pat. No. 7,010,482. The present invention relates to vector quantization (VQ) in speech coding systems using waveform interpolation. In recent years, there has been increasing interest in achieving toll-quality speech coding at rates of 4 kbps and below. Currently, there is an ongoing 4 kbps standardization effort conducted by an international standards body (The International Telecommunications Union-Telecommunication (ITU-T) Standardization Sector). The expanding variety of emerging applications for speech coding, such as third generation wireless networks and Low Earth Orbit (LEO) systems, is motivating increased research efforts. The speech quality produced by waveform coders such as code-excited linear prediction (CELP) coders degrades rapidly at rates below 5 kbps; see B. S. Atal, and M. R. Schroeder, (1984) “Stochastic Coding of Speech at Very Low Bit Rate”, On the other hand, parametric coders, such as: the waveform-interpolative (WI) coder, the sinusoidal-transform coder (STC), and the multiband-excitation (MBE) coder, produce good quality at low rates but they do not achieve toll quality; see Y. Shoham, Commonly in WI coding, the similarity between successive rapidly evolving waveform (REW) magnitudes is exploited by downsampling and interpolation and by constrained bit allocation; see W. B. Kleijn, and J. Haagen, (1995), The present invention describes novel methods that enhance the performance of the WI coder, and allows for better coding efficiency improving on the above 1999 Gottesman and Gersho procedure. The present invention incorporates analysis-by-synthesis (AbS) for parameter estimation, offers higher temporal and spectral resolution for the REW, and more efficient quantization of the slowly-evolving waveform (SEW). In particular, the present invention proposes a novel efficient parametric representation of the REW magnitude, an efficient paradigm for AbS predictive VQ of the REW parameter sequence, and dual-predictive AbS quantization of the SEW. More particularly, the invention provides a method for interpolative coding input signals, the signals decomposed into or composed of a slowly evolving waveform and a rapidly evolving waveform having a magnitude, the method incorporating at least one various, preferably combinations of the following steps or can include all of the steps: (a) AbS VQ of the REW; (b) parametrizing the magnitude of the REW; (c) incorporating temporal weighting in the AbS VQ of the REW; (d) incorporating spectral weighting in the AbS VQ of the REW; (e) applying a filter to a vector quantizer codebook in the analysis-by-synthesis vector-quantization of the rapidly evolving waveform whereby to add self correlation to the codebook vectors; and (f) using a coder in which a plurality of bits therein are allocated to the rapidly evolving waveform magnitude. In addition, one can combine AbS quantization of the slowly evolving waveform with any or all of the foregoing parameters. The new method achieves a substantial reduction in the REW bit rate and the EWI achieves very close to toll quality, at least under clean speech conditions. These and other features, aspects, and advantages of the present invention will become better understood with regard to the following detailed description, appended claims, and accompanying drawings. In very low bit rate WI coding, the relation between the SEW and the REW magnitudes was exploited by computing the magnitude of one as the unity complement of the other; see W. B. Kleijn, and J. Haagen, (1995), “A Speech Coder Based on Decomposition of Characteristic Waveforms”, IEEE ICASSP'95, pp. 508-511; W. B. Kleijn, and J. Haagen, (1995), “Waveform Interpolation for Coding and Synthesis”, in Also, since the sequence of SEW magnitude evolves slowly, successive SEWs exhibit similarity, offering opportunities for redundancy removal. Additional forms of redundancy that may be exploited for coding efficiency are: (a) for a fixed SEW/REW decomposition filter, the mean SEW magnitude increases with the pitch period and (b) the similarity between successive SEWs, also increases with the pitch period. In this work we introduce a novel “dual-predictive” AbS paradigm for quantizing the SEW magnitude that optimally exploits the information about the current quantized REW, the past quantized SEW, and the pitch, in order to predict the current SEW. Introduction to REW Quantization The REW represents the rapidly changing unvoiced attribute of speech. Commonly in WI systems, the REW is quantized on a waveform by waveform base. Hence, for low rate WI systems having long frame size, and a large number of waveforms per frame, the relative bitrate required for the REW becomes significantly excessive. For example, consider a potential 2 kbps system which uses a 240 sample frame, 12 waveforms per frame, and which quantizes the SEW by alternating bit allocation of 3 bit and 1 bit per waveform. The REW bitrate is then 24 bit per frame, or 800 kbps which is 40% of the total bitrate. This example demonstrates the need for a more efficient REW quantization. Efficient REW quantization can benefit from two observations: (1) the REW magnitude is typically an increasing function of the frequency, which suggests that an efficient parametric representation may be used; (2) one can observe a similarity between successive REW magnitude spectra, which may suggest a potential gain by employing predictive VQ on a group of adjacent REWs. The next two sections propose REW parametric representation, and its respective VQ. REW Parametric Representation Direct quantization of the REW magnitude is a variable dimension quantization problem, which may result in spending bits and computational effort on perceptually irrelevant information. A simple and practical way to obtain a reduced, and fixed, dimension representation of the REW is with a linear combination of basis functions, such as orthonormal polynomials; see W. B. Kleijn, Y. Shoham, D. Sen, and R. Haagen, (1996), A simple and practical way for parametric representation of the REW is, for example, by a parametric linear combination of basis functions, such as polynomials with parametric coefficients, namely: One can observe the similarity between successive REW magnitude spectra, which may suggest a potential gain by VQ of a set of successive REWs. R _{1}(ω),R _{2}(ω), . . . ,R _{M}(ω)]^{T} (4)
and the VQ output is an index, j, which determines a quantized parameter vector, {circumflex over (ξ)}: {circumflex over (ξ)}=[{circumflex over (ξ)} _{1},{circumflex over (ξ)}_{2}, . . . ,{circumflex over (ξ)}_{M}]^{T} (5)
which parametrically determines a vector of quantized spectra: (ω)={circumflex over (R)} {circumflex over ((ω,{circumflex over (ξ)})=[{circumflex over (R)}R)}(ω,{circumflex over (ξ)}_{1}),{circumflex over (R)}(ω,{circumflex over (ξ)}_{2}), . . . ,{circumflex over (R)}(ω,{circumflex over (ξ)}_{M})]^{T} (6)
The encoder searches, in the parameter codebook C _{q}(ξ), for the parameter vector which minimizes the distortion:
A. Quantization Using Orthonormal Functions Orthonormal functions, such as polynomials, may be used for efficient quantization of the REW; see W. B. Kleijn, et al., (1996),
B. Piecewise Linear Parametric Representation In order to have a simple representation that is computationally efficient and avoids excessive memory requirements, we model the two dimensional surface by a piecewise linear parametric representation. Therefore, we introduce a set of N uniformly spaced spectra, {{circumflex over (R)}(ω,{circumflex over (ξ)} C. Weighted Distortion Quantization Commonly in speech coding, the magnitude is quantized using weighted distortion measure. In this case the quantized REW parameter is then given by:
D. Weighted Distortion—Piecewise Linear Parametric Representation Again, for practical considerations assume that the parametric representation is piecewise linear, and may be represented by a set of N spectra, {{circumflex over (R)}(ω,{circumflex over (ξ)} This section presents the AbS VQ paradigm for the REW parameter. The first presentation is a system which quantizes the REW parameter by employing spectral based AbS. Then simplified systems, which apply AbS to the REW parameter, are presented. A. REW Parameter Quantization by Magnitude AbS VQ The novel Analysis-by-Synthesis (AbS) REW parameter VQ technique is illustrated in The scheme incorporates both spectral weighting and temporal weighting. The spectral weighting is used for the distortion between each pair of input and the quantized spectra. In order to improve SEW/REW mixing, particularly in mixed voiced and unvoiced speech segments, and to increase speech crispness, especially for plosives and onsets, temporal weighting is incorporated in the AbS REW VQ. The temporal weighting is a monotonic function of the temporal gain. Two codebooks are used, and each codebook has an associated predictor coefficient, P A sequence of quantized parameter, such as ĉ(k), is formed by concatenating successive quantized vectors, such as {ĉ B. Simplified REW Parameter AbS VQ The above scheme maps each quantized parameter to coefficient vector, which is used to compute the spectral distortion. To reduce complexity, such mapping, and spectral distortion computation, which contribute to the complexity of the scheme, may be eliminated by using the simplified scheme described below. For a high rate, and a smooth representation surface {circumflex over (R)}(ω,ξ), the total distortion is equal to the sum of modeling distortion and quantization distortion: B.1. Simplified REW Parameter AbS VQ—Non Weighted Distortion B.2. Simplified REW Parameter AbS VQ—Weighted Distortion The simplified quantization scheme is improved to incorporate spectral and temporal weightings, as illustrated in In order to exploit the information about the pitch and voicing level, the possible pitch range was partitioned into six subintervals, and the REW parameter range into three. Also, eighteen codebooks were generated, one for each pair of pitch range and unvoicing range. Each codebook has associated two mean vectors, and two diagonal prediction matrices. To improve the coder robustness and the synthesis smoothness, the cluster used for the training of each codebook overlaps with those of the codebooks for neighboring ranges. Since each quantized target vector may have a different value of the removed mean, the quantized mean is added temporarily to the filter memory after the state update, and the next quantized vector's mean is subtracted from it before filtering is performed. The output weighted SNR, and the mean-removed weighted SNR, of the scheme are illustrated in Examples for the two predictors for three REW parameter ranges are illustrated in Bit Allocation The bit allocation for the 2.8 kbps EWI coder is given in Table 1. The frame length is 20 ms, and ten waveforms are extracted per frame. The line spectral frequencies (LSFs) are coded using predictive MSVQ, having two stages of 10 bit each, a 2-bit increase compared to the past version of our code; see O. Gottesman and A. Gersho, (1999),
Subjective Results A subjective A/B test was conducted to compare the 2.8 kbps EWI coder of this invention to G.723.1. The test data included 24 modified intermediate reference system (M-IRS) filtered speech sentences, 12 of which are of female speakers, and 12 of male speakers; see ITU-T, (1996), “Recommendation P.830, Subjective Performance Assessment of Telephone Band and Wideband Digital Codecs”, Annex D, ITU, Geneva. Twelve listeners participated in the test. The test results, listed in Table 2 and Table 3, indicate that the subjective quality of the 2.8 kbps EWI exceeds that of G.723.1 at 5.3 kbps, and it is slightly better than that of G.723.1 at 6.3 kbps. The EWI preference is higher for male than for female speakers.
Table 2 shows the results of subjective A/B test for comparison between the 2.8 kbps EWI coder to 5.3 kbps G.723.1. With 95% certainty the result lies within +/−5.53%.
Table 3 shows the results of subjective A/B test for comparison between the 2.8 kbps EWI coder to 6.3 kbps G.723.1. With 95% certainty the result lies within +/−5.59%. It should, of course, be noted that while the present invention has been described in terms of an illustrative embodiment, other arrangements will be apparent to those of ordinary skills in the art. For example; While in the disclosed embodiment in While in the disclosed embodiment was related to waveform interpolative speech coding, in other arrangements it may be used in other coding schemes. While in the disclosed embodiment temporal weighting, and/or spectral weighting are described, they are optional, and in other arrangements any or both of them may not be used. While in the disclosed embodiment switch prediction having two predictors is described, in other arrangements no switch, or more than two predictor choice may be used. While in the disclosed embodiment illustrated in While in the disclosed embodiment the pitch range and/or the voicing parameter values were partitioned into subranges, and codebooks were used for each subrange, this may be viewed as optional, and in other arrangements any or all of such subranges may not be used, or other number or type of subranges may be used. While in the disclosed embodiment describes prediction matrices were diagonal, in other arrangements non diagonal prediction matrices may be used. The following references are each incorporated herein by reference: - B. S. Atal, and M. R. Schroeder, “Stochastic Coding of Speech at Very Low Bit Rate”,
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